Anything from Waves is great. I particularly enjoy C4, LinMB, MaxxBass, and L2. As far as a rule of thumb for stacking them goes, you should always have L2 as your final plugin and stack effects after compressors/multiband compressors such as stacking MaxxBass after C4 or LinMB. I'm no professional here, but I have been reading up on and playing with Waves plugins for a couple years now.

I was quite happy with my QCD player because it was resource friendly. But now that I have Adapt-X running :Waves Q10 Paragraphic EQ-> Waves MaxxBass-> Waves S1 Shuffler-> Antares Tube-> Waves L2 (Waves version 4.0, is there a newer one?)I have an average 25% consumption of CPU power on my Athlon XP 2100+.

But I love my new plug-ins They rock.

I have a question. What's the internal resolution of Waves Plug-ins? Can I pass a 32 bit signal through them until L2 dithers and truncates to 16 bit using Adapt-X (I know Adapt-X supports transfer of 32 bit resolution between the elements in the chain).

I was quite happy with my QCD player because it was resource friendly. But now that I have Adapt-X running :Waves Q10 Paragraphic EQ-> Waves MaxxBass-> Waves S1 Shuffler-> Antares Tube-> Waves L2 (Waves version 4.0, is there a newer one?)I have an average 25% consumption of CPU power on my Athlon XP 2100+.

But I love my new plug-ins They rock.

I have a question. What's the internal resolution of Waves Plug-ins? Can I pass a 32 bit signal through them until L2 dithers and truncates to 16 bit using Adapt-X (I know Adapt-X supports transfer of 32 bit resolution between the elements in the chain).

LOL. Subscribing though to the fact that the waves plugins are the best, I have to confess that I prefer the loudness & tone controls on a good preamp instead.

They do entirely different things and used for different purposes. Those plug-ins do noise shaping. Like L2 does dithering or Maxx Bass alters the bass tones. The REQ (Renaissance Equalizer) is a non-linear equalizer. There're analog equalizer that does that but not the "decent" ones I could ever afford. I have to say I never enjoyed my music so deep, it is worth the whopping 25 % CPU time consuption.

BTW these plug-ins are also implemented in hardware AFAIK but cost a lot of money and are huge, although probably doing a lot better than the software plug-in Anyway that's far beyond my budget. But you can't achieve the same effect with a regular decent pre-amp.

This post has been edited by atici: May 14 2003, 17:10

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The object of mankind lies in its highest individuals.One must have chaos in oneself to be able to give birth to a dancing star.

L2's nothing but a final-stage compressor, right? It's a combination dither and "MAKE IT LOUD" device from what I've seen. Since the plugin chain is probably re-quantizing to 16-bit for each component anyway, there's no need to use its 24bit->16bit dither, and it won't yield any appreciable resolution improvement. I used to have it on my Adapt-X plugin chain just for the hell of it, but then I took it off when I decided it did more harm than good. It was just chopping off more dynamic range on already over-compressed music.

I haven't had good luck with MaxxBass, either. After reading the manual, I concluded that it's designed for speakers with poor low-frequency response; generating upper harmonics allows previously unhearable bass to be heard. So, since my headphones already go down to 20Hz, there's not much need for it. Then I tried using it on some cheap PC speakers, according to the manual: find where the frequency response drop-off is, and set that to be where the MaxxBass harmonic generator starts, and optionally disable "original bass". But to my ears, it made no benefit. Low-frequency "jungle" basslines were still inaudible, and rock'n'roll bass drums just sounded muddier. As a low-frequency "boost", maybe MaxxBass is more effective than an equalizer, but as a low-frequency "extender", I was unimpressed.

However, don't let my grouchy grumblings stop you from enjoying your plugins! Tinkering with them is certainly a lot of fun, and it teaches you about audio applications as well. Reading their well-written manuals (all available online for free, if you use a Google search) will teach you even more. Plus, if you've shown more patience than I did, then you probably have found settings that make a real improvement in your music.

Since SometimesWarrior touched the subject of re-quantization between any plugin in the chain, I would like to mention this:

Some people rave about iZotope Ozone because it does all the processing internally in 64bits precision - no need to re-quantize to 16 bits between each plugin in the chain. It's only one plugin, with the chain of modules inside (Tupperware inside Tupperware, to use Monty's analogy)

To make things clearer, I will quote the iZotope page (therefore, take it with a grain of salt)

QUOTE

it just sounds better because, once the signal goes in, it's all 64-bit processing until it comes out. It's technically impossible to get this sound quality and resolution by chaining plug-ins together.

I personally don't use it, because the audio processing I do with Waves here is more subtle, and only requires one or two plugins to get to the results I want. (I don't use them as Winamp DSP, either)

Regards;

Roberto.

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Get up-to-date binaries of Lame, AAC, Vorbis and much more at RareWares:http://www.rarewares.org

I have already been reading about the plug-ins I use actually (I found them on google as you said). Before I explain what L2 does, I want to say I am not a puritan about exact audio reproduction but rather I listen to my music I enjoy it. That does not mean I like distorting my music though, tonal shaping/alteration should be kept within reasonable limits. So I turn the plugins I use off/on and adjust the volume so that the gain some plug-ins apply won't make difference and see how I like it better. And I read/learn/know about them and only use them if I believe in their principles (like dither example).

And as far as I read about these plug-ins, they seem to be based on rather sound principles. And I have to say, Waves or BombFactory or other professional plugins should not be confused with $20 plugins (like DFX) out there. Most of the stuff designed for the masses is hyped using the same idiotic terminology (check HDCD or XM site for instance). They use vague terms that would impress average Joe. On the other hand the manuals of these plugins are written in professional language and tells exactly what the plugin is capable of doing and what it cannot achieve. And if these plugins were all about snake oil then these companies would be out of business already.

The difference, as far as I could describe, in music when plugins are on are the additional warmth and tonal richness which are of course subjective. But my perception tells me the music has become so much enjoyable now. Some might say it's a placebo effect and it's about loving your equipment (and in this case your software) and the attachment to the new set of buttons/bells & whistles to tinker with...

QUOTE (SometimesWarrior @ May 14 2003 - 06:59 PM)

L2's nothing but a final-stage compressor, right? It's a combination dither and "MAKE IT LOUD" device from what I've seen.

Now about L2. It's a peak limiter and a noise shaping & dithering tool. The peak limiting is interesting, L2 adds gain to most of the song in the level you desire but for the peaks of the song that actually should not determine the average song volume it attenuates the volume without being noticeable using some sort of lookahead and predicting. It is some kind of local replaygaining.

Peak Limiting:

QUOTE

For mastering purposes, the peak level of the processed signal would normally be set to 0dB, or just below 0dB. Because a typical digital audio file of music contains many high intensity, short duration peaks, simple normalization of the file may still result in a low average signal level. Using the L2-Ultramaximizer however, it is generally possible to significantly increase the average signal level of a typical audio file without introducing any audible side effects...ABOUT MAXIMUM LEVELThe maximum level of a digital signal is governed by the highest peak in the file. Simple normalization finds the highest peak, then raises the entire signal so that this peak is at the maximum value. However, many of these peaks may be of very short duration and can usually be reduced in level by several dBs with minimal audible side effects. Those familiar with digital editing systems may even have proved this for themselves by "redrawing" some trouble-some peaks by hand. By transparently controlling these peaks, the entire level of the file can be raised several more dB than by simple normalization resulting in a higher average signal level. The L2-Ultramaximizer avoids the possibility of overshoot by utilizing a lookahead technique that allows the system to anticipate and reshape signal peaks in a way that produces the bare minimum of audible artifacts. Because there is no possibility of overshoot, L2 can be used with absolute confidence in situations where brickwall limiting is important.

Dithering and Noise Shaping:

QUOTE

ABOUT DITHER AND NOISE SHAPINGDithering and Noise shaping are two independent, but complementary, techniques to improve the perceived quality of sound after it has been requantized.As will be explained here in some detail, each technique is responsible for the improvement of a different subjective quality of the noise imposed by re-quantization. Therefore, each can be used separately to improve that specific quality.

Dithering is done in order to change the character of the quantization noise to more closely resemble analog hiss, rather than digital quantization noise. The main effect of dithering is to reduce (or, in case of type1, virtually eliminate) all correlation between the quantization noise and the original signal, thus reducing (eliminating) non-linear distortion typical of digital quantization noise. The dithering process 'exchanges' these distortions for a steadier analog-hiss quality signal. Noise shaping is done in order to optimize the distribution of overall noise energy across the spectrum. This optimization is according to the ear's sensitivity. This means a decrease in noise (whether distortion or hiss) in the ear's sensitive areas (1 to 6kHz), is 'exchanged' for an increase of noise in less sensitive areas (above 15kHz, toward Nyquist). Hopefully this has helped you see that in both techniques, the issue is about 'exchanging' the character and frequency content of noise (hiss & distortion) according to subjective criteria.

QUOTE (SometimesWarrior @ May 14 2003 - 06:59 PM)

Plus, if you've shown more patience than I did, then you probably have found settings that make a real improvement in your music.

Yes I really think I did.

This post has been edited by atici: May 15 2003, 04:23

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The object of mankind lies in its highest individuals.One must have chaos in oneself to be able to give birth to a dancing star.